[Freeswitch-users] DTMF detection giving problems

SamyGo govoiper at gmail.com
Fri Aug 10 12:30:55 MSD 2012


is that ms ? God bless the poor callers !! Goodness. wait I'll be back.

On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  7 ms seems like a very short timeout to have?****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Från:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *För *SamyGo
> *Skickat:* den 10 augusti 2012 10:11
> *Till:* FreeSWITCH Users Help
> *Ämne:* Re: [Freeswitch-users] DTMF detection giving problems****
>
> ** **
>
> No Ivan , I've no issue at all with DTMFs ever. Like I said there is an
> Asterisk server which used torun the same IVRs and on same setup and never
> got such a problem.****
>
> But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in
> between. Calls are coming in OK, I intermittently face this issue where
> user presses DTMFs and those digits are not recognized by
> FS-PlayAndGetDigits() ****
>
> I've  lines like this in my LUA****
>
> ** **
>
>  choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "",
> "[0-9]")****
>
>  freeswitch.consoleLog("INFO","User Entered "..choice.." for this
> Survey\n")****
>
> ** **
>
> *Logs for the calls without dtmf here:*****
>
> ** **
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280****
>
> ** **
>
> --------------------- Another Example---------------------****
>
> ** **
>
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
> ** **
>
> --------------------- Another Example---------------------****
>
> ** **
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
> ** **
>
> ** **
>
> ** **
>
> *Log for successful DTMF collected is as follows:*****
>
> ** **
>
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]****
>
> [INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
> -------****
>
> ** **
>
> ** **
>
> So I can see a pattern here. But I can't figure out this behavior. Is it
> because I've 1 try only ! and after just once playing of file PAGD() breaks
> no matter it gets DTMF or not !?****
>
> ** **
>
> Regards,****
>
> Sammy****
>
> ** **
>
> ** **
>
> On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com>
> wrote:****
>
> Hello Sammy.****
>
> should check whether the operator is not blocking dtmf****
>
> I had a situation - in the same statement it worked, on the other - not
> processed incoming dtmf ****
>
> ** **
>
> 2012/8/10 SamyGo <govoiper at gmail.com>****
>
> Hello All,****
>
> ** **
>
> I've a small lua script which is just an IVR. Calls are landing on
> Asterisk and then they are relayed over to FS. The problem is that the IVR
> don't always detects what user pressed. What I've observed is that the DTMF
> threshold is very high and some people just press slowly/quickly and the
> tone duration gets low and FS don't detect anything. ****
>
> ** **
>
> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
> also liberal-dtmf set in my sofia external.xml profile file which somewhat
> got me better results but still DTMFs miss sometimes.****
>
> ** **
>
> Please note that the asterisk on which the calls are landing have never
> faced this issue before. Asterisk has PRI from which it receives the calls
> and then relay to desired FS server.****
>
> ** **
>
> ** **
>
> Regards.****
>
> Sammy****
>
> ** **
>
> ** **
>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
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>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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>
> Join Us At ClueCon - Aug 7-9, 2012
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> !DSPAM:5024c05932762100017234! ****
>
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>
> 
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